Voice over Internet Protocol (VoIP) is a form of Internet telephony. In an Internet telephone call, an analogue voice signal is converted to digital format and the signal is compressed into Internet protocol (IP) packets for transmission over the Internet.
Like other data networks, VoIP transfers the data packets via different ways, but not always fast enough, which may lead to transmission leakage or distortion.
This technology allows a group of people to make telephone calls using a broadband Internet connection instead of a regular (or analogue) phone line. As explained above, one significant difference to traditional telephony is the technique used to transmit voice data. Instead of the classic voice switching technology, VoIP uses data packet transmission to send information and directs only the “data packets” of voice information to their destination.
This technology often uses the infrastructure of an already existing network by sharing it with other communication services. To this extent, since the VoIP network has to deal with the complexity of numerous network measures, like data packet protection or compression, it is necessary to ensure the secure traffic of the information against attackers.
The introduction of such measures to the VoIP network complicates several aspects of VoIP performance and in particular the dynamic port traffic and call setup procedures. In addition, to avoid intruders compromising the VoIP network, some other layers of defense are necessary to protect the voice traffic. Such layers of defense are factors that need to be included at the IP level, like encryption techniques, security signature and the like. Thus, it is mandatory to protect the traffic network.
However, these additional factors cause an excessive amount of latency in the VoIP packet delivery and degrade the Quality of Service (QoS) that is a fundamental point to the operation of the VoIP network. This can lead to a degraded voice quality together with an unceasing tradeoff adjustment between security and voice quality. Unfortunately, due to the time-critical nature of the VoIP, the implementation of corrective measures considerably degrades the quality of the voice transmission. These measures impact the inherent latency of the VoIP protocol (i.e., the time it takes for a voice transmission to go from its source to its destination).
Without entering into the details of the VoIP protocol, it is important to mention that the latency is a genuine constraint that impacts both the VoIP performance and the transfer quality. Ideally, the latency value has to be set as low as possible to successfully emulate the QoS of today's telephones.
Since the data packets “travel” across the Internet, there exists also a potential risk of dropouts and an undesirable generation of clipping during a VoIP voice transmission. Such effects, combined with the genuine constraints (as explained herein above), make it difficult to provide acceptable voice quality during a call.
An end user (particularly if he/she is a speaker) is completely unaware of the voice quality level when speaking. At worst, the speaker's voice may be completely garbled to the receiving parties as the speaker continues speaking with no indication of any voice reception problem. The speaker would only become aware of a problem when he/she stops speaking and is told by others of the poor quality during the call. This is particularly a problem for conference calls as the presenter is unaware that he/she can't be understood, leaving him/her speaking for many minutes while receiving no indication of a poor voice quality.
There exist several real-time monitoring systems available on the market today such as the WinEyeQ call monitor/protocol analyser from Touchstone Inc. This type of system provides a solution for monitoring call performance in terms of successful/failed calls. However, this type of system does not provide to the end user a real-time indication of the voice quality during the call.
US20020167937A1 to Goodman describes a method for testing voice call quality in a VoIP network. The method includes enabling a communications device connected to the VoIP network to answer a test call received over the VoIP network by playing a voice file, generating a test call over the VoIP network to the communications device, and measuring voice call listening quality from the voice file played by the communications device. While this solution provides a measure of the voice quality on a test call, it does not provide to the speaker a real-time indication of the voice quality during a live call.
Thus, there is a need for a system and method that informs in real-time the caller and the listeners in a Voice over Internet Protocol network of the quality of the voice communication.